[Asterisk-Users] Re: Grandstream Budgetone SIP registration fails

[David J Carter]

Tony,

Have a look here http://www.codepipe.com/id25.htm these are my working
examples.

I have 6 GS phones. The GS set-up's are from extersion 8002 onwards in
sip.conf.

Regards

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Tony
Mountifield
Sent: 06 March 2004 21:04
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Grandstream Budgetone SIP registration
fails


In article <1078595645.9560.49.camel at Senecio>,
Jean-Marc V. Liotier <asterisk-users at lists.digium.com> wrote:
> Someone on the list certainly has a working setup with Asterisk and
> Grandstream Budgetone phones, I would be grateful if their SIP
> configuration was posted to the list. Quite unexpectedly I found no
> complete example of such working setup on the Web, maybe because it was
> so simple that no one thought that posting it would be useful to anyone.
> One I get mine working I shall post the parameters !

Well my sip.conf looks like this:

----------------------------------------------------------------------------
--
;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context=from-sip-external       ; send unknown SIP callers to this context
allow=ulaw
allow=ilbc

;
; Tony's phone
;
[2000]
type=friend
username=2000
secret=password
host=dynamic
context=from-sip-internal
mailbox=2000
callerid=2000
dtmfmode=info

;
; Rachel's phone
;
[2001]
type=friend
username=2001
secret=password
host=dynamic
context=from-sip-internal
mailbox=2001
callerid=2001
dtmfmode=info
----------------------------------------------------------------------------
--

Then in the admin interface for Tony's phone I have the following:

IP address: dynamic from DHCP
SIP server: IP of Asterisk server
Outbound proxy: empty
SIP User ID: 2000
Authenticate ID: 2000
Auth password: password
Vocoder choices (in order): PCMU, PCMA, then others
....
SIP user ID is phone number: Yes
SIP Registration: Yes
Clear reg on reboot: No
Reg expiration: 3
Early dial: No
....
Local SIP port: 5060
Local RTP port: 5004
Use random port: No
NAT Traversal: No
....
Send DTMF: Via SIP INFO
....

I think that's all the likely relevant ones.

Hope this helps
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
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